[..] |
ASTERISK-14000: [patch] Greetings are stored as IMAP messages even when imapgreetings=no |
ASTERISK-14001: this is a test |
ASTERISK-14002: pthread_rwlock_timedrdlock breaks OpenBSD build. |
ASTERISK-14003: Last digit missing when dialing out to pstn and echotraining=yes or echotraining=xx |
ASTERISK-14004: Trunk registration / Auth user |
ASTERISK-14005: [patch] Add config options to change some dial features |
ASTERISK-14006: Dial() within AEL Macros does not allow use of 'h' extension |
ASTERISK-14007: Segfault Asterisk 1.4.24.1 |
ASTERISK-14008: Attaching msg0001.WAV instead of msg0000.WAV |
ASTERISK-14009: externip is ignored for Audio unless localnet is defined |
ASTERISK-14010: Asterisk causes kernel panic while outound calling |
ASTERISK-14011: Asterisk gets unresponsive and shows maximum channels used, even call volume is low |
ASTERISK-14012: ilbc calls causing core dumps |
ASTERISK-14013: [patch] chan_vpb fails to catch exception on 1.4 |
ASTERISK-14014: Monitor in Asterisk does not record all calls and also recordings are not complete according to the complete actual duration |
ASTERISK-14015: FIELDQTY not function correctly with escape characters from variables. |
ASTERISK-14016: [patch] Add new command VMSayName() |
ASTERISK-14017: [patch] erros messages astobj2.c INTERNAL_OBJ |
ASTERISK-14018: [patch] "misdn show config" segfaults asterisk, if no MSN lists |
ASTERISK-14019: no iax trunking on 1.6.1 svn |
ASTERISK-14020: 200 OK is not accepted when SIP INFO in early dialog |
ASTERISK-14021: don't warn, when pipe is used in app_system |
ASTERISK-14022: Crash on Dial when using S() option to hangup call after n secs. |
ASTERISK-14023: [patch] Compilation issue on Solaris due to ast_channel_unref |
ASTERISK-14024: doc/core-en_US.xml missed an ending tag |
ASTERISK-14025: [patch] include contexts based on time causes asterisk to loop on startup |
ASTERISK-14026: segfault during attended transfer of an automatically redirected call |
ASTERISK-14027: Asyncagi break missing |
ASTERISK-14028: invalid value in Remote-Party-ID (RPID) |
ASTERISK-14029: [patch] app_osplookup fails to build with libosptk3-dev 3.4.2-1: ‘OSPT_CERT’ undeclared |
ASTERISK-14030: [patch] invalid XML syntax in doc of 'Queue' |
ASTERISK-14031: [patch] Missing error message if indications.conf is not found |
ASTERISK-14032: Segfault with chan_woomera |
ASTERISK-14033: [patch] [regression] #0013747 not fixed for local channel (Indications are not passed from old peer to new peer during masquerad |
ASTERISK-14034: [patch] SIP Response 410 incorrectly mapped to Hangupcause 1, should be 22 |
ASTERISK-14035: [patch] Invalid SDP connection information (c=) parsing leading to one way audio |
ASTERISK-14036: Multiple parking lots don't work |
ASTERISK-14037: Rewrite Skinny sub handling |
ASTERISK-14038: [patch] Change readq locking |
ASTERISK-14039: hpec echo cancellation not working in Asterisk 1.4.24/dahdi-latest |
ASTERISK-14040: [patch] CoreStatus Data |
ASTERISK-14041: [patch] CoreShowChannels Response does not honor actionid |
ASTERISK-14042: [patch] Opening voice channel on FastStartAcknowledged before Answer. Remove H245inSetupOptions for better capability. |
ASTERISK-14043: [patch] Diameter implementation of Asterisk CDR. |
ASTERISK-14044: LAGRQ Warnings in 1.6.1 |
ASTERISK-14045: [patch] Introduce better parsing for the register line |
ASTERISK-14046: Asterisk's not handling BYE sip-tls messages |
ASTERISK-14047: DIALPLAN_EXISTS does not check for Caller ID |
ASTERISK-14048: TDM400P card unable to get the hook status when system started |
ASTERISK-14049: indications.c:149 playtones_generator: Can't generate that much data! |
ASTERISK-14050: [patch] Asterisk loses SIP phones, possible deadlock, 1.6.1.0 |
ASTERISK-14051: [patch] Patch that makes app_dial set channel variable DIVERTED_BY on 302 response |
ASTERISK-14052: [patch] Patch that makes chan_sip check if the forward domain is itself on a 302 response |
ASTERISK-14053: [patch] armv5tel: ast_expr2.c op_func.c implicit declerations |
ASTERISK-14054: [patch] live_ast: edit asterisk.conf instead of creating it |
ASTERISK-14055: tab completion sefault when using with CLI aliases |
ASTERISK-14056: cdr_odbc ont work in some ver of 1.6.0.6 to 1.6.1 (and next ?) |
ASTERISK-14057: [patch] Language handling for numbers, dates, etc is misbehaving when utilizing sub-regional languages |
ASTERISK-14058: [patch] Bad locking logic on res_config_mysql.c |
ASTERISK-14059: DTMF is not working correctly for cell phones. |
ASTERISK-14060: [patch] Functions INC and DEC |
ASTERISK-14061: Asynchronous Javascript Asterisk Manager (AJAM) , not able to log in in internet explorer |
ASTERISK-14062: [patch] silent 'ringing' for branched calls |
ASTERISK-14063: [patch] Add 16khz WAV support (format_wav16.c) |
ASTERISK-14064: [patch] Asterisk Addons 1.6.1.0 does not always honor DESTDIR |
ASTERISK-14065: Setting the 'T' option for Meetme does not send MeetmeTalking events to the manager |
ASTERISK-14066: [patch] Sign problem calculating timestamp for iax frame leads to no audio on the receiving peer. |
ASTERISK-14067: chan_sip sets PRIREDIRECTREASON incorrectly for reason no-answer |
ASTERISK-14068: [patch] ast_say_number_full_ur function in main/say.c |
ASTERISK-14069: stop the MOH since asterisk knows that channel is ringing |
ASTERISK-14070: [patch] ignore both DTMF BEGIN and END from RTP when not in RFC2833 mode |
ASTERISK-14071: [patch] chan_mobile.c:1683 sco_accept: incoming audio connection for pvt without owner |
ASTERISK-14072: [patch] Function ODBC_FETCH returns more than one row |
ASTERISK-14073: DTMF's are not sent to the bridge channel if they are used by any built-in or dynamic feature |
ASTERISK-14074: Crash on attended transfer |
ASTERISK-14075: Call via mobile randomly hangs or |
ASTERISK-14076: Override Early Media |
ASTERISK-14077: [patch] Fetching a NULL value from database returns "NULL" string |
ASTERISK-14078: [patch] Data truncated in ooh323_request because of sizeof incorrect usage |
ASTERISK-14079: Chan_mobile does not have reload command |
ASTERISK-14080: [patch] NT over PtMP for BRI |
ASTERISK-14081: [patch] pri set to NULL in pri_message in function dump_facility |
ASTERISK-14082: MeetMe Fails to Authenticate |
ASTERISK-14083: [patch] Moh class set in the dialplan is ignored with realtime moh |
ASTERISK-14084: [patch] registration fails if multiple peers are specified in sip.conf |
ASTERISK-14085: "dahdi.conf" misused in conf files |
ASTERISK-14086: [patch] Vietnamese support for Voicemail |
ASTERISK-14087: [patch] span numbers in pri debug / error messages |
ASTERISK-14088: [patch] Voicemail password changes not working as expected |
ASTERISK-14089: [patch] Blank FORWARD_CONTEXT is not ignored |
ASTERISK-14090: [patch] hints with 2+ devices that include ONHOLD are often set wrong |
ASTERISK-14091: Core Dump on 1.4.24 - local_pvt_destroy (pvt=0x193cbc00) at chan_local.c:159 |
ASTERISK-14092: If the database loses connection for a few seconds, no cdr gets written ever |
ASTERISK-14093: fail to bridge channels |
ASTERISK-14094: [patch] Email notification of voicemail segfaults Asterisk |
ASTERISK-14095: voice mail early media |
ASTERISK-14096: [patch] Add Queue Reset ability to Reset all members and all Numbers |
ASTERISK-14097: [patch] Cannot make or receive mobile calls. |
ASTERISK-14098: [patch] Possible crash in pbx_spool.c - operation on previously freed structure |
ASTERISK-14099: [patch] CHANNEL(transfercapability) not documented |
ASTERISK-14100: huge memory leak - sip_alloc |
ASTERISK-14101: [patch] chan_mobile.c:1038 mbl_read: read error 107 |
ASTERISK-14102: [patch] Early media bridged from caller to callee allows free calls |
ASTERISK-14103: [patch] Realtime SIP not working in current SVN |
ASTERISK-14104: STRFTIME returns incorrect time |
ASTERISK-14105: Segfault on Transfer |
ASTERISK-14106: MOH not closing calls correctly |
ASTERISK-14107: [patch] undefined LOG_WARNING and LOG_NOTICE in main/alaw.c and main/ulaw.c |
ASTERISK-14108: [patch] Support for event keep-alive for Linksys NAT pinhole |
ASTERISK-14109: Channel bridge |
ASTERISK-14110: [patch] Broadcasting CDR &| Manager Events via UDP |
ASTERISK-14111: [patch] Asterisk crash when logging out from a manager session. |
ASTERISK-14112: [patch] When using lots of queues and agents, asterisk will coredump |
ASTERISK-14113: [patch] chan_sip random deadlock |
ASTERISK-14114: [patch] digit timeout problem with 1.4 pbx.c rev 193119 |
ASTERISK-14115: [patch] astcanary: race when asterisk daemonizes |
ASTERISK-14116: [patch] Correct connection time in 'realtime mysql status' |
ASTERISK-14117: Behaviour when dealing with Diversion that leads back to Asterisk |
ASTERISK-14118: DTMF occurs in every calls without user press keyboard of telephone |
ASTERISK-14119: [patch] ast_channel_free might double unlock channels lock |
ASTERISK-14120: 1.6.1/DAHDI: No outgoing analogue calls possible |
ASTERISK-14121: [patch] Directed pickup : picker picks own channel instead of called party's channel |
ASTERISK-14122: [patch] SIP allowguest defaults to yes with 'make samples' |
ASTERISK-14123: [patch] Registration Deadlock between Asterisk and Polycom Soundpoint IP 450 |
ASTERISK-14124: [patch] At the end of an attended transfered, on hangup, Asterisk crashes. |
ASTERISK-14125: [patch] Asterisk Manager API Action Originate / OriginateResponse |
ASTERISK-14126: [patch] Random loss of sound when using G.729 |
ASTERISK-14127: Routing Extensions between 2 asterisk servers in 2 directions fails with "482 Loop Detected" |
ASTERISK-14128: AstXML Documentation System Has No Ability To Include Documentation Sections |
ASTERISK-14129: [patch] Abort by memory allocator, possibly in moh_files_generator |
ASTERISK-14130: Caller identifier problem |
ASTERISK-14131: [patch] parsing of sip register lines is broken |
ASTERISK-14132: AstXML documentation system can't handle nested enum lists |
ASTERISK-14133: autofill problem when coming back from paused on realtime |
ASTERISK-14134: [patch] calling ConfBridge() with no timing source causes segfault |
ASTERISK-14135: Identificar chamadas fax ou de voz eletronica |
ASTERISK-14136: Millions of "We're Zap/4-2, not", than crash |
ASTERISK-14137: crash in voicemail when more than maxmsg already stored in folder |
ASTERISK-14138: [patch] SIP device cycles between Available for 25 seconds and Unavailable for the programmed re-registration period, maybe an h |
ASTERISK-14139: [patch] Video support in SIP channel driver appears to be totally broken |
ASTERISK-14140: Abandoned queue calls won't show on CDR |
ASTERISK-14141: out of bounds crash and core dump |
ASTERISK-14142: Segmentation fault with incoming calls |
ASTERISK-14143: [patch] meetme fails looking for conf-getconfno |
ASTERISK-14144: does AGI ported "send manager events" function ? |
ASTERISK-14145: sometimes : dial_exec_full: Unable to create channel of type 'IAX2' (cause 20 - Unknown) |
ASTERISK-14146: [patch] Incoming DTMF causes "Cannot handle frames in 2 format" error, call dies |
ASTERISK-14147: [patch] Voicemail "greetings only" feature |
ASTERISK-14148: Null fields do not get stored as null in mysql |
ASTERISK-14149: asterisk does not stream using currrent versions of mpg123 |
ASTERISK-14150: [patch] Dynamic parking lots |
ASTERISK-14151: SIP stops working with multiple REGISTER statements in sip.conf |
ASTERISK-14152: Hangup()-command in dialplan does not clear the channel |
ASTERISK-14153: DNS host name resolution in iax.conf |
ASTERISK-14154: [patch] Deadlock on chan_sip |
ASTERISK-14155: Errors on manager.c when DEBUG_THREADS is enabled |
ASTERISK-14156: Asterisk rejects t38 reinvite when using local channels |
ASTERISK-14157: [patch] Small Patch for 1.4 and 1.6 for report Original Position on ENTERQUEUE event. |
ASTERISK-14158: [patch] Invalid syntax with two ampersands in if block |
ASTERISK-14159: [patch] Refactor queue member properties and add functions to retrieve/modify them |
ASTERISK-14160: No audio on SIP RE-INVITE connecting with AllWorx PBX |
ASTERISK-14161: segmentation fault in Asterisk caused by fun_odbc |
ASTERISK-14162: [patch] Deadlock On One-legged Transfer [SIP / REPLACES] (Call Pickup) |
ASTERISK-14163: [patch] 64 bit system channel name uniqueness |
ASTERISK-14164: Interoperability 64bit problem SPARC crash Asterisk |
ASTERISK-14165: [patch] SIP Call-Limit |
ASTERISK-14166: [patch] When lastms was added to realtime_update_peer in chan_sip the necessary ldap changes were not made |
ASTERISK-14167: [patch] Annoying "Unknown RTP codec 126 received" messages confuse people |
ASTERISK-14168: [patch] Message: "Unable to handle indication 3" |
ASTERISK-14169: [patch] MGCP Business Phone Packages patch |
ASTERISK-14170: DAHDI takes considerably longer than Zaptel to complete call |
ASTERISK-14171: RT voicemail storage : save the message itself in the DB |
ASTERISK-14172: [patch] Parked calls do not ring back after timeout |
ASTERISK-14173: Voicemail support |
ASTERISK-14174: The 't' in the dial command allows the transfer but disconnects the legs! |
ASTERISK-14175: SIP Registrations ignore the "fromdomain" setting. |
ASTERISK-14176: [patch] 'core stop when convenient' causes segfault |
ASTERISK-14177: func_odbc is broken in 1.4.25 |
ASTERISK-14178: [patch] Add ability to use extension state as well as device state when adding queue memebers |
ASTERISK-14179: When building with uClibc, configure script mistakenly assumes iconv is always available |
ASTERISK-14180: [patch] MySQL ENUM Type Not Detected |
ASTERISK-14181: hints : DEVSTATE UNAVAILABLE presence state of "terminated" not interpreted correctly on Grandstream devices |
ASTERISK-14182: [patch] Issue a warning, if trying to run gosub outside dialplan |
ASTERISK-14183: Error rate for 'asterisk.c: Accept returned -1: Too many open files' is not throttled. |
ASTERISK-14184: [patch] cant unload and reload app_minivm.so |
ASTERISK-14185: [patch] v.110 dialin support for ISDN channels |
ASTERISK-14186: "sip show subscriptions" shows false state |
ASTERISK-14187: Segmentation fault. Strange sound generate in channal when onhook(without segmentation fault). |
ASTERISK-14188: 1.6.0.9 does not build on Darwin PowerPC |
ASTERISK-14189: [patch] No unique identifier for CDR |
ASTERISK-14190: [patch] file convert leaks input file descriptor |
ASTERISK-14191: [patch] T.38 invite does not always comply with RFC 2327 |
ASTERISK-14192: [patch] Attended Transfers are not working |
ASTERISK-14193: [patch] trunk no longer compiles in dev-mode |
ASTERISK-14194: [patch] Patch to allow tone-list as argument to ReadExten |
ASTERISK-14195: [patch] Asterisk crash when recording busy or unavailable message while using ODBC voicemail storage |
ASTERISK-14196: [patch] ReadExten returns TIMEOUT in cases where it should return OK or INVALID |
ASTERISK-14197: [patch] #exec script can't access manager on first asterisk load |
ASTERISK-14198: Ability to assign which nic card or mac address to use for each trunk in sip.conf |
ASTERISK-14199: IAX trunk only comes up as trunk in one direction only, unless you unload and load chan_iax |
ASTERISK-14200: [patch] segfault in local_devicestate() in chan_local.c |
ASTERISK-14201: [patch] Gosub AGI command is not being registered. |
ASTERISK-14202: [patch] [SIP realtime] "sip reload" makes UNREACHABLE users behind NAT |
ASTERISK-14203: double free or corruption (!prev) in moh_files_generator |
ASTERISK-14204: [patch] Asterisk crash when using ODBC to insert record on table that does not exist |
ASTERISK-14205: G729 Decoders not releasing after mixmonitor ends |
ASTERISK-14206: [patch] adding queue member from asterisk console with state_interface parameter not working |
ASTERISK-14207: VOICEMAIL : I've tried a lot but mailing is not working... |
ASTERISK-14208: [patch] Certain console helps leaves prompt in bold mode (bright white) |
ASTERISK-14209: IAX2 immediately retries after a failed registration, causing a flood of failed registrations |
ASTERISK-14210: Parsing of register statements in sip.conf fails when transport is used without extension |
ASTERISK-14211: Suggested change in sip.conf comments |
ASTERISK-14212: Dropping frame since I'm still dialing on DAHDi/1-1... |
ASTERISK-14213: Inband DTMF Double digits being sent. |
ASTERISK-14214: [patch] The CUT function does not show the leading field-separators until it finds a value. |
ASTERISK-14215: make install: build_tools/mkpkgconfig: 21: [[: not found |
ASTERISK-14216: [patch] Asterisk 1.6.1.0 crashes with a core dump at random occassions |
ASTERISK-14217: [patch] asterisk lock in sipsock_read for several seconds and drop sip packets |
ASTERISK-14218: AEL Macro Records [s] for DST in CDR |
ASTERISK-14219: Crash in INVITE with replaces |
ASTERISK-14220: after transfer negotiation, releasing doesnt work , both channels are still up....forever |
ASTERISK-14221: Nice to have setting to point to sox/soxmix |
ASTERISK-14222: deadlock in res_timing_pthread and chan_sip do_monitor/rettransmit |
ASTERISK-14223: recent commit on SVN 1.4 branch fails to compile |
ASTERISK-14224: IAX2 choppy audio with MozIAX, only 20secs |
ASTERISK-14225: segfault after "restart when convenient" using dundi |
ASTERISK-14226: attended transfer of an automatically redirected call does not work |
ASTERISK-14227: [patch] New asterisk function REPLACE |
ASTERISK-14228: [patch] mp3 playback not working via AGI "STREAM FILE", but "EXEC PLAYBACK" works |
ASTERISK-14229: make config on slackware |
ASTERISK-14230: [patch] Simplify main/Makefile removing the list of objects to compile |
ASTERISK-14231: DEVICE_STATE() always returns 0 (Unknown) |
ASTERISK-14232: [patch] Disable automatic subscribe accept |
ASTERISK-14233: [patch] Buddies are always auto-registered when processing the roster |
ASTERISK-14234: I can not connect to asterisk console |
ASTERISK-14235: SIP Realtime qualify goes crazy till Asterisk crash |
ASTERISK-14236: [patch] XML parse errors related to buffer size, and not handled properly |
ASTERISK-14237: MixMonitor stops after transfer from queue |
ASTERISK-14238: [patch] If dahdi timing interface is not registered, when we try to unload res_timing_dahdi asterisk crash. |
ASTERISK-14239: [patch] Make chan_mobile signal progress when the mobile phone has accepted the dialed number |
ASTERISK-14240: [patch] Fix sms support |
ASTERISK-14241: [patch] Added emaildatelocale option for variable VM_DATE |
ASTERISK-14242: invalid SIP/NOTIFY header (Multiple values in single-value header Event and Content-Type) |
ASTERISK-14243: IAX2 realtime username confusion |
ASTERISK-14244: Realtime IAX2 crash |
ASTERISK-14245: [patch] log does not indicate which function is missing closing parenthesis |
ASTERISK-14246: wrong sip port number display |
ASTERISK-14247: [patch] Move static docs to the new AstXML form |
ASTERISK-14248: [patch] Multiple Groups Not working |
ASTERISK-14249: crash - address out of bounds |
ASTERISK-14250: [patch] internal timing not working if HAVE_ZAPTEL (dahdi_compat issue) |
ASTERISK-14251: Incorrect path in make_version_h |
ASTERISK-14252: app_voicemail.so could not be loaded |
ASTERISK-14253: [patch] segfault (sig 6) |
ASTERISK-14254: Audio lost in following prompts after DTMF pressed during Background() |
ASTERISK-14255: [patch] headset will not stop ringing |
ASTERISK-14256: MixMonitor is not releasing the file handle on the recorded file |
ASTERISK-14257: Asterisk 1.6.2.0 does not compile in mac os X 10.5.8 |
ASTERISK-14258: [patch] periodic-announce-frequency suggestion |
ASTERISK-14259: [patch] RFC3261 Via-header branches not done right (section |
ASTERISK-14260: Asterisk crashed |
ASTERISK-14261: ReceiveFAX does not produce CED tones, fax reception times out on DAHDI channels |
ASTERISK-14262: Calling over TAPI hangs up the line if trunk responses with "SIP/2.0 403 Forbidden, no minutes left" |
ASTERISK-14263: x86_64 relocation R_X86_64_32 |
ASTERISK-14264: [patch] memory leak in asterisk some bug fixing and removing Redundant condition |
ASTERISK-14265: [patch] Bad handling of 488 answer to re-invite |
ASTERISK-14266: [patch] BASE64_DECODE() adds garbage end end of decoded string |
ASTERISK-14267: [patch] german time (20:01:00 oh clock) is announced wrong |
ASTERISK-14268: [patch] SIP peers remain present in the channel's memory after rename (and probably removal) |
ASTERISK-14269: [patch] apps/app_festival.c does not compile for PPC target |
ASTERISK-14270: [patch] SMS FIX for motorola phones |
ASTERISK-14271: [patch] BRI Network side PTMP mode. Support USA NI1 protocol. |
ASTERISK-14272: [patch] add initial support for AT+CUSD command |
ASTERISK-14273: music on hold digit=X non-functional |
ASTERISK-14274: LibSS7 not working when linkset is set to a value > 2 |
ASTERISK-14275: [patch] Asterisk 1.6.0.10 crashes randomly |
ASTERISK-14276: [patch] CLI NOTIFY always tries to use UDP, even if the peer is connected via TCP |
ASTERISK-14277: version 1.6.1.1 problem with audio playback |
ASTERISK-14278: [patch] reworked chan_ooh323 |
ASTERISK-14279: RTCP (SR) message is not scheduled to send backto UAC in case 2 UACs are the same codec. |
ASTERISK-14280: Macro() returns to wrong extension |
ASTERISK-14281: [patch] Aborted inbound trunk call to FXO analog port causes internal extensions (SIP or DAHDI) to ring forever. |
ASTERISK-14282: [patch] Billsec is zero when playing a file in the Dial function |
ASTERISK-14283: Segmentation fault on Re-INVITE when t38pt_udptl=no |
ASTERISK-14284: [regression] Received invalid event that had no device IE |
ASTERISK-14285: Meetme ignores 'q' option |
ASTERISK-14286: Error loading module 'chan_h323.so': undefined symbol: _ZNK7PObject7CompareERKS_ |
ASTERISK-14287: Registrations persist after removal and sip reload |
ASTERISK-14288: TDM call over IAX trunk gives choppy audio, SIP call over same IAX trunk sounds fine. |
ASTERISK-14289: Build of 1.6.0.10 fails on Darwin |
ASTERISK-14290: Build of 1.6.1.1 fails on Darwin |
ASTERISK-14291: Calling with Local |
ASTERISK-14292: native asterisk format moh playback sounds garbled |
ASTERISK-14293: Typo in ast_db_gettree |
ASTERISK-14294: Make second rfcomm connection |
ASTERISK-14295: new_find_extension arguments in wrong order |
ASTERISK-14296: Mixmonitor generates small files then the actual call |
ASTERISK-14297: Asterisk crash after hangup 2 bluetooth cellphone channels same time |
ASTERISK-14298: Busydetect failure on unanswered calls because I am having the exact same issue. (reopen issue 0003813) |
ASTERISK-14299: [patch] ExternalIVR does not handle arguments in a consistant manner |
ASTERISK-14300: After 20 mins with heaviy traffic it blocks |
ASTERISK-14301: DTMF duration absurdly lon gwhen passed from SIP to DAHDI |
ASTERISK-14302: No possible adjust txgain over DAHDI channel |
ASTERISK-14303: [patch] incorrect comparation in ast_monitor_change_fname() leads to deletion of recorded files |
ASTERISK-14304: [patch] Seg fault in chan_local - local_pvt_destroy |
ASTERISK-14305: [patch] Segfault after Manager Bridge |
ASTERISK-14306: Asterisk can not get channel lock when receiving BYE message which results in 503 error message |
ASTERISK-14307: busylevel does not work with realtime |
ASTERISK-14308: [patch] CUT() returns empty string for fields other than the 1st |
ASTERISK-14309: [patch] Patch to make DUNDi load balance across hosts with the same weight |
ASTERISK-14310: [patch] DUNDILOOKUP() does not accept comma as argument separator |
ASTERISK-14311: DISA fails. |
ASTERISK-14312: res_agi.c:2299: error: '__WORDSIZE' undeclared |
ASTERISK-14313: Fax Session does not stop when channel hangups |
ASTERISK-14314: Asterisk crash in ast_channel functions with dumps |
ASTERISK-14315: [patch] Using CHANNEL function from ZOMBIE channel stops Asterisk |
ASTERISK-14316: [patch] Log message does not match conditional check |
ASTERISK-14317: [patch] Recorded files get deleted before mixing if the call was transfered |
ASTERISK-14318: [patch] add FILE_STORAGE to Voicemail Build Options |
ASTERISK-14319: Mutliple parking lot |
ASTERISK-14320: Need to expand number of available flags in chan_iax2 |
ASTERISK-14321: deadlock during call bridge, mutex lock |
ASTERISK-14322: Drawn out and static audio on inbound iax2 calls |
ASTERISK-14323: AstApplicationData when called with several arguments escapes comma with backslash |
ASTERISK-14324: [patch] Deadlock when performing directed pickup |
ASTERISK-14325: parsing of sip register lines is still broken |
ASTERISK-14326: [patch] lock in sip_tcp_helper_thread |
ASTERISK-14327: [patch] appdocs dtd installed to ASTVARLIBDIR instead of ASTDATADIR |
ASTERISK-14328: [patch] SIP deadlock in 1.4 revision 199472 |
ASTERISK-14329: [patch] TW is not an ISO Language Code |
ASTERISK-14330: Unanswered attended transfers get the voicemail of the transferrer. Not the intended extension. |
ASTERISK-14331: When dialing a feature code, devstate function returns NOT_INUSE |
ASTERISK-14332: Deadlock in do_monitor() of chan_sip |
ASTERISK-14333: Segfault on agent login, with manager, agi and pgsql realtime |
ASTERISK-14334: On upgrade from version 1.6.0.6 to 1.6.1.0 Error: "No matching peer found" |
ASTERISK-14335: MixMonitor - utils.c: write() returned error: Broken pipe |
ASTERISK-14336: app_fax does not compile with iaxmodem 1.2.0 |
ASTERISK-14337: After a few thousand calls, or at random, Asterisk stops receiving events from the network |
ASTERISK-14338: [patch] Documentation fix for CLI usage of update2 |
ASTERISK-14339: [patch] Remove unneeded define for Solaris |
ASTERISK-14340: [patch] Remove unused defines |
ASTERISK-14341: [patch] Several "recieved" typos in source files |
ASTERISK-14342: [patch] AST-2009-001 breaks IAX2 RFC5456 compliance - Timestamps in POKE/PONG zero in 2 of 4 Bytes |
ASTERISK-14343: [patch] log message output is truncated |
ASTERISK-14344: 1.6.2.0_beta3 locks up in certain SIP scenario |
ASTERISK-14345: File Permissions On Voicemails Left To Multiple Recipients Incorrect |
ASTERISK-14346: ChanIsAvail is returning unavailable when the device is actually available |
ASTERISK-14347: recent commit on svn 1.4 chan_sip failed to load |
ASTERISK-14348: Call failed to go through, [...] instead of excuting next extension |
ASTERISK-14349: Sometimes SEGFAULTS when TDM800P FXS port goes off hook |
ASTERISK-14350: Queues Disconnect Callers |
ASTERISK-14351: [patch] Repeatedly building the asterisk directory repeatedly downloads the sounds files |
ASTERISK-14352: [patch] Russian syntax. app_voicemail forgets to say "messages" when there are no new and old messages in the mailbox. |
ASTERISK-14353: [patch] app_followme does not set correct language/inherit from calling channel for Local/xxxxx channels it creates |
ASTERISK-14354: [patch] T38 reinvite started from Asterisk |
ASTERISK-14355: [patch] VMWI not sent after every OnHook, only when messages changes from None to Some, or Some to None |
ASTERISK-14356: [patch] realtime mysql status always shows connected for 0 seconds/no connection |
ASTERISK-14357: SIP option (SIP_OPT_ flag) is not handled correctly |
ASTERISK-14358: [patch] segfault in iax2_hangup Asterisk revision 201600 |
ASTERISK-14359: [patch] Crash in do_monitor() in chan_dahdi.c |
ASTERISK-14360: Crash when trying to log via ast_log |
ASTERISK-14361: Park() application options not working as documented |
ASTERISK-14362: [patch] TDM400P FXS port, doesn't ring correctly when mwisendtype=rpas |
ASTERISK-14363: SIP TLS URIs are not consistent with RFC 3261 |
ASTERISK-14364: Unsupported SDP media type in offer |
ASTERISK-14365: SIP clients and "internal_timing" not working when silence suppression enabled |
ASTERISK-14366: [patch] Milliwatt() is off by -11dbm |
ASTERISK-14367: Option g don't work if caller hangup call |
ASTERISK-14368: [patch] no audio with SIP call to ISDN PRI, if neither Progress or Proceeding are received. |
ASTERISK-14369: Option U() functionality is not equivalent to application documentation |
ASTERISK-14370: [patch] 1.6.1.1: Memory handling error in main/pbx.c (pbx_extension_helper) |
ASTERISK-14371: After update to 1.6.0.10 ReceiveFAX doesn't work. |
ASTERISK-14372: [patch] Memory leak in func_audiohookinherit.c |
ASTERISK-14373: Dialling Fast on SIP (484) Does not match Dialplan |
ASTERISK-14374: app_queue segfault |
ASTERISK-14375: [patch] segfault in action_coreshowchannels() at manager.c |
ASTERISK-14376: hints not working for agents |
ASTERISK-14377: rxfax crash asterisk |
ASTERISK-14378: /include/asterisk/lock.h:531 __ast_pthread_mutex_unlock: app_mixmonitor.c line 277 (mixmonitor_thread): Error releasing mutex: O |
ASTERISK-14379: [patch] SIP Unique Channel Name |
ASTERISK-14380: [patch] new Record() option |
ASTERISK-14381: [patch] Cannot find XML documentation under non linux platforms |
ASTERISK-14382: [patch] Session timer is not activated |
ASTERISK-14383: [patch] Unrequired Debug Message |
ASTERISK-14384: Early media causes "Asked to transmit frame type 64, while native formats is 0x8 (alaw)(8)......" |
ASTERISK-14385: Asterisk crashes at approx 70-80 calls per second |
ASTERISK-14386: Multiple m=video or m=audio lines cause a ip port number mismatch. |
ASTERISK-14387: False Answer Supervision when setting parameter nocallsetup=yes |
ASTERISK-14388: Template system overriding |
ASTERISK-14389: 1.4.24.1 deadlock in devicestate.c ast_device_state_engine_init |
ASTERISK-14390: [patch] Mapping of extension state to device state is incorrect |
ASTERISK-14391: crash: "chan_sip.c", lineno=3910 |
ASTERISK-14392: crash: "frame.c", lineno=505 |
ASTERISK-14393: No voice on PRI calls with asterisk 1.4.25 & 26 |
ASTERISK-14394: Repeated DTMF |
ASTERISK-14395: Asterisk ignores incoming H323 overlap |
ASTERISK-14396: Move Addons into Asterisk branch |
ASTERISK-14397: [patch] No audio on calls from asterisk sip phones to nortel set until dtmf from sip phone |
ASTERISK-14398: [patch] Serious problem in pattern matching |
ASTERISK-14399: G729 codec configuration |
ASTERISK-14400: Gosub-Local Variables in AEL2 Cannot be set in a Queue with Local Channels |
ASTERISK-14401: System crash |
ASTERISK-14402: [patch] No way to pass CFLAGS to non-module objects |
ASTERISK-14403: MixMonitor stops recording when call is transferred |
ASTERISK-14404: [patch] app_voicemail hangup trying to play vm-helpexit-full file because doesn't exists |
ASTERISK-14405: [patch] asterisk crashes in voicemail |
ASTERISK-14406: ast_load_realtime doesn't return correct values |
ASTERISK-14407: [patch] MP3 files not playing in Asterisk 1.6.1.1 + Addons 1.6.1.0 |
ASTERISK-14408: crash / call forwarding possibly due to loop detection - 1.4.26rc4 |
ASTERISK-14409: [patch] URIENCODE() throws a warning when passed an empty string |
ASTERISK-14410: Revision 203638 seems to break the update of hints |
ASTERISK-14411: [patch] Crash when performing directed pickup |
ASTERISK-14412: [patch] Asterisk cannot handle SIP 183 "Session Progress" if no SDP is contained in it |
ASTERISK-14413: mixmonitor mutex freed more times than locked |
ASTERISK-14414: Transport (TCP/UDP) not used correctly |
ASTERISK-14415: sip notify always sent in UDP |
ASTERISK-14416: [patch] dummy-select: better listing of modules |
ASTERISK-14417: [patch] If function MEETME_INFO() is run on a conference with no participants, return is not numeric |
ASTERISK-14418: Unable to make ISDN PRI calls, due to PRI-Exclusive=0 for PRI-CPE. |
ASTERISK-14419: [patch] Unable to make ISDN PRI calls after upgrade. |
ASTERISK-14420: There are no faxes! |
ASTERISK-14421: DTMF tones missing randomly |
ASTERISK-14422: asterisk freezes with 100% CPU after receiving a fax when -p option (realtime priority) is used |
ASTERISK-14423: Recent IAX2 changes in 1.4 SVN spam /var/log/asterisk/messages |
ASTERISK-14424: MixMonitor stop call bridging |
ASTERISK-14425: Support for Polycom VVX 1500 video |
ASTERISK-14426: [patch] mISDN rejects calls - NO FREE CHAN IN STACK |
ASTERISK-14427: [regression] parking c-e-p is missing, so timed-out parked call never returns to parker |
ASTERISK-14428: Asterisk Crashed When made a attended Transfer |
ASTERISK-14429: Crash becouse don't check null return... |
ASTERISK-14430: Asterisk crashes with ChanSpy between a SIP and Agent channel |
ASTERISK-14431: Problem with queue |
ASTERISK-14432: [patch] It crashes often daily and always in the same function |
ASTERISK-14433: [patch] crash in bridging api |
ASTERISK-14434: rtpkeepalive option doesn't always work |
ASTERISK-14435: [patch] Add option and description to chan_dahdi.conf.sample |
ASTERISK-14436: DTMF's are not sent to the bridge channel if they are used by any built-in or dynamic feature |
ASTERISK-14437: OPTIONS Messages Flood - Memory Crash |
ASTERISK-14438: cdr_custom produces incorrect csv format for clid |
ASTERISK-14439: "Segmentation fault " caused by reinvite |
ASTERISK-14440: random segfault |
ASTERISK-14441: [patch] Race condition in cdr_syslog.c (SVN Revision 205561) |
ASTERISK-14442: AGI Script dialparties.agi returned error: Broken Pipe |
ASTERISK-14443: callerid(num) is wrong when username is missing |
ASTERISK-14444: [patch] Issues cross-compiling dahdi-tools 2.2.0 |
ASTERISK-14445: [patch] Building dahdi-linux-2.1.0.4 as cross-compilation works, but 2.2.0 fails with wrong architecture type |
ASTERISK-14446: [patch] Not all fixes from #14849 are committed |
ASTERISK-14447: [patch] [regression] iaxclient (tested with Zoiper) registered to asterisk shows devicestate Unavailable instead Not-InUse |
ASTERISK-14448: Memory leak in file.c |
ASTERISK-14449: Segmentation Fault loading res_fax.so module |
ASTERISK-14450: FollowMe plays wrong sound files |
ASTERISK-14451: [patch] [branch] RTMP support in Asterisk |
ASTERISK-14452: 183 response although progressinband=never |
ASTERISK-14453: callerid in Canada does not work for Voicetronix Openswitch12 |
ASTERISK-14454: automon feature logs nothing under -vvv |
ASTERISK-14455: [patch] mISDN rejects incoming calls (reopened) |
ASTERISK-14456: DTMFs don't work |
ASTERISK-14457: wcte12xp0: Missed interrupt. when disable echocanceller |
ASTERISK-14458: [patch] Voicemail.conf gets overwritten when using Realtime for voicemail |
ASTERISK-14459: [patch] contrib/scripts/meetme.sql doesn't contain all fields |
ASTERISK-14460: Unused structure member in app_queue |
ASTERISK-14461: [patch] Asterisk runs over end of buffer reading manager input over HTTP and segfaults |
ASTERISK-14462: reference count is not decreased properly in app_queue |
ASTERISK-14463: asterisk crashes once per day |
ASTERISK-14464: [patch] Manager events for Skinny |
ASTERISK-14465: app_voicemail cannot connect to IMAP server |
ASTERISK-14466: Setting CDR(userfield) fails when called from inside a Macro initiated from a feature. |
ASTERISK-14467: [patch] Patch for new feature: sip show channels concise |
ASTERISK-14468: [patch] G726 Codec has choppy audio on Version 1.6.1 |
ASTERISK-14469: Asterisk crashing in cmd voicemail |
ASTERISK-14470: Asterisk 1.4.26rc5 (revision 202945 ) deadlock in monitor.c "Locked Here: db.c line 151 (ast_db_put)" |
ASTERISK-14471: Queue members penalties issue |
ASTERISK-14472: [patch] Segfault - Perhaps in sig_analog.c |
ASTERISK-14473: Asterisk 1.4.26-rc6 revision 202945 locked in cdr.c line 1060 (post_cdr) |
ASTERISK-14474: Not able to install Asterisk on OpenSUSE 11.1 |
ASTERISK-14475: [patch] Deadlock between ast_cel_report_event and ast_do_masquerade |
ASTERISK-14476: TIMEOUT(absolute) return negative value |
ASTERISK-14477: [wishlist] Goto should have parameter arguments just like Gosub |
ASTERISK-14478: [patch] Update to coding guidelines |
ASTERISK-14479: [patch] Note about using unixODBC |
ASTERISK-14480: [patch] memory leak in func_realtime |
ASTERISK-14481: HDLC Bad FCS (8) on PRI and call was failed |
ASTERISK-14482: iax.conf, IP-based access control |
ASTERISK-14483: Millions of "ERROR[24278]: channel.c:2046 __ast_read: ast_read() called with no recorded file descriptor." messages spamming CLI |
ASTERISK-14484: module reload causes asterisk to crash |
ASTERISK-14485: Can not using T.38 origination with app_fax/spandsp |
ASTERISK-14486: [patch] chan_dahdi deadlock heavy incomming traffic |
ASTERISK-14487: [patch] sig_pri.c Trying To Set NPI When There Is No Destination Number |
ASTERISK-14488: [patch] Channel Unlocked Two Times |
ASTERISK-14489: switch => does not work with variables |
ASTERISK-14490: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 and later the system crash |
ASTERISK-14491: DISA not working with addons-1.6.1.1-rc2 |
ASTERISK-14492: [patch] Add voicefile and dtmf options to res/res_agi.c |
ASTERISK-14493: Getting one way audio after blind transfer from a SIP trunk call. |
ASTERISK-14494: WARNING[9078]: file.c:677 ast_readaudio_callback: Failed to write frame |
ASTERISK-14495: "sip show peers" does not show realtime peers |
ASTERISK-14496: WARNING[5251]: chan_dahdi.c:1669 dahdi_set_hook: DAHDI hook failed returned -1 (trying 1): Device or resource busy |
ASTERISK-14497: Asterisk 1.6.1.1 fails to cross-compile (ppc-target) |
ASTERISK-14498: Segfault When Trying To Reconnect To MySQL Server Using ODBC |
ASTERISK-14499: [patch] Multi-tenant parking broken in 1.6.1.1 - does not allocate to designated parking spaces |
ASTERISK-14500: [patch] SIP OPTIONS qualify message forever |
ASTERISK-14501: [patch] Register request line contains wrong address when domain and registrar host differ |
ASTERISK-14502: [patch] chan_sip behaves like crazy if there is not route to destination |
ASTERISK-14503: [patch] chan_iax2 sends command RINGING in answer state |
ASTERISK-14504: app_waituntil does not wait on 1.6.1 |
ASTERISK-14505: Segfault in SVN revision 207360 chan_sip.c |
ASTERISK-14506: Segfault in SVN revision 207360 chan_sip.c in in __ast_pthread_mutex_lock |
ASTERISK-14507: Not passing audio on a sip call in and out on the same peer |
ASTERISK-14508: Phone disconnects Asterisk Trunk Revision 207723 |
ASTERISK-14509: [patch] Initialization error in chan_mobile |
ASTERISK-14510: [patch] SIP_BODY function to get a body part of a SIP message |
ASTERISK-14511: Reason Header / Answered Elsewhere Flag Not Used by Queues |
ASTERISK-14512: NoCDR generating records when called in 'h' exntesion |
ASTERISK-14513: Symbol exports for strlcat and strlcpy |
ASTERISK-14514: [patch] Gosub() dequotes once more than Macro() |
ASTERISK-14515: Bridged channels doesn't free after call termination |
ASTERISK-14516: stateinterface in queues.conf makes the member show "Invalid" |
ASTERISK-14517: Crash on chan_local |
ASTERISK-14518: [patch] Parked Call timeout fails to return call back to originating extension |
ASTERISK-14519: Compilation fails on Opensolaris |
ASTERISK-14520: CPU at 100%, no calls going through |
ASTERISK-14521: [patch] install_prereq does not work on Debian 5 |
ASTERISK-14522: [patch] 'received' typos in trunk, in 6 files |
ASTERISK-14523: [patch] p->peerauth is always empty in transmit_register() |
ASTERISK-14524: core show hints |
ASTERISK-14525: SendFAX not working with T.38 |
ASTERISK-14526: Simultaneously multiple call dialing and if anybody will answer, playback an message |
ASTERISK-14527: [patch] Add busy detection |
ASTERISK-14528: [patch] add support for circular searching for free devices in a group of phones |
ASTERISK-14529: [patch] #exec strips too many leading and trailing quotes |
ASTERISK-14530: RTCP jitter incorrect |
ASTERISK-14531: [patch] Failure to negotiate T.38 |
ASTERISK-14532: Segfault in chan_sccp when registering Cisco 7931 |
ASTERISK-14533: [patch] crash in chanspy on hangup - locked mutex '&chanspy_ds.lock' |
ASTERISK-14534: [patch] Rtptimeout not honored when sip channels are bridged |
ASTERISK-14535: deny caller transfer to the extension |
ASTERISK-14536: [patch] Comfort noise frame with f->data NULL but f->datalen 160 |
ASTERISK-14537: AMI doesn't handle COLP correctly when bridging IAX and SIP channels after an attended transfer |
ASTERISK-14538: One way audio after attended transfer |
ASTERISK-14539: Inband DTMF begin event not fired correctly |
ASTERISK-14540: Possible crash in astobj2 |
ASTERISK-14541: [patch] Queue member considered available when paused, causing high weight queue to block low weight queue |
ASTERISK-14542: [patch] fix leaks and cppcheck warning |
ASTERISK-14543: [patch] app_background fails to reliably detect DTMF when echocan module is enabled |
ASTERISK-14544: Is not setting the CALLERID(num) from the dahdi-channels.conf information |
ASTERISK-14545: IAX does not bind to multiple virtual IP's correctly |
ASTERISK-14546: Crash in autoservice (locking) |
ASTERISK-14547: [patch] fix spelling for typos, mainly in comments. |
ASTERISK-14548: [patch] all codecs allowed, but textsupport=no crashes on T140RED enabled call |
ASTERISK-14549: [patch] Session-Expires Parse Error |
ASTERISK-14550: SIP crash on ACK |
ASTERISK-14551: Have another queue ringing strategy to ring all members with penalties between min/max |
ASTERISK-14552: chan_dahdi parser assigns channel 1 to wrong context |
ASTERISK-14553: Dtmf Pound /Hash Key does not work properly when using via a local channel |
ASTERISK-14554: CDR(dest) records as s when using ael Macros |
ASTERISK-14555: app_fax.c is not compiling under OpenBSD |
ASTERISK-14556: [patch] set NOISY_BUILD to yes when compiling under devmode |
ASTERISK-14557: 'app_voicemail.so' could not be loaded - undefined symbol: ast_smdi_interface_find |
ASTERISK-14558: [patch] WARNING[23025]: channel.c:952 __ast_queue_frame: Exceptionally long voice queue length queuing to Local |
ASTERISK-14559: Transfer to parking lot replies '202 Accepted' rather than '480 Temporarily Unavailable' when lot is full |
ASTERISK-14560: T.38 re-INVITE received after T.38 already negotiated fails |
ASTERISK-14561: Frequent SIP registrations cause firewall packet drop cycle |
ASTERISK-14562: DAHDI restart causes multiple 'dahdi_pri_error: Can't destroy call 0!' on console |
ASTERISK-14563: Changes to CALLERID don't get propagated to CDR(clid) |
ASTERISK-14564: [patch] maintenance support broke compatiblity with old libpri |
ASTERISK-14565: realtime function does not return pair when column data value is null |
ASTERISK-14566: random crashes |
ASTERISK-14567: [patch] crash in LOCAL() if Gosub stack is allocated but empty |
ASTERISK-14568: [patch] Channel not locked when it should in local_attended_transfer |
ASTERISK-14569: [patch] session-expires default timer wrong |
ASTERISK-14570: DISA still not working. |
ASTERISK-14571: [patch] 'h' extension never reach in a macro |
ASTERISK-14572: res_ais, communication ok, but wrong state send and receive. |
ASTERISK-14573: [patch] "core show codecs" segfaults on Solaris. |
ASTERISK-14574: [patch] Prepending to a voicemail on forward causes locked sip channel and large file filling disk space |
ASTERISK-14575: [patch] Vietnamese support for SayNumber() function. |
ASTERISK-14576: [patch] Asterisk runs out of sockets |
ASTERISK-14577: Asterisk crash when attempting outgoing mobile call. |
ASTERISK-14578: Problem with IP extention for incomming calls |
ASTERISK-14579: Asterisk crushes if I hangup call immediately after I make it from google talk to my Asterisk |
ASTERISK-14580: Queue autopause |
ASTERISK-14581: [patch] Module 'chan_dahdi.so' could not be loaded. |
ASTERISK-14582: [patch] [branch] Implement standard XMPP Jingle in Asterisk |
ASTERISK-14583: "skype show users" in chan_skype Beta does not work |
ASTERISK-14584: Skype 1.4.0.11 Make error |
ASTERISK-14585: [patch] Deadlock after peer answeres queue call |
ASTERISK-14586: [patch] event AgentComplete - fields differ by reason |
ASTERISK-14587: [patch] compilation fails in systems without index() |
ASTERISK-14588: [patch] Send CallerID on the QueueCallerAbandon manager event |
ASTERISK-14589: [patch] Fix for Sonus DTMF issues |
ASTERISK-14590: Update docs to state that canreinvite does NOT stop Asterisk from issuing reinvites for non-direct-media purposes. |
ASTERISK-14591: Monitor not working, terminates dialplan |
ASTERISK-14592: Steady crasy every few hours under load of 10-15 concurrent calls |
ASTERISK-14593: T38 Faxing failing on 1.6.1 svn |
ASTERISK-14594: SVN view not available |
ASTERISK-14595: snmp SubAgent does not return channel count for SIP tech |
ASTERISK-14596: [patch] Return from Park still using | as delimiter |
ASTERISK-14597: [patch] Missing new-message notification for urgent messages |
ASTERISK-14598: [patch] Dialplan starts execution before call is accepted |
ASTERISK-14599: [patch] res_calendar.c does not compile in devmode |
ASTERISK-14600: main/file.c updated as part of addons merge to replace connected line information for each file playback |
ASTERISK-14601: Snom phones occasionally fail to register in 1.6.1.3rc1 |
ASTERISK-14602: ChanSpy "whisper" is broken in 1.4.26 |
ASTERISK-14603: CDR dynamic fields in postgresql |
ASTERISK-14604: [patch] QUEUE_MEMBER_LIST() returns member names instead of interfaces |
ASTERISK-14605: Asterisk eats up all the processor and prints the same message in the screen |
ASTERISK-14606: Parked Call Back fails due to use of old parameter format |
ASTERISK-14607: LOGGER WARNING : error executing after rotate |
ASTERISK-14608: AGENTACCEPTDTMF is incorrectly spelled as AGENTACCEPTDMTF in code to recognize channel variables. |
ASTERISK-14609: asterisks crashes when trying to load skype channel |
ASTERISK-14610: [patch] Checking IMAP folder for voicemail only gets messages in the same context and extension |
ASTERISK-14611: [patch] Stuck channel using FEATD_MF if caller hangs up at the right time |
ASTERISK-14612: ? in register= string breaks registration |
ASTERISK-14613: [patch] Length of the parameters using an Async Originate |
ASTERISK-14614: [patch] o option not working |
ASTERISK-14615: [patch] ringt formatting cleanup, possible bug in distintive ring |
ASTERISK-14616: [patch] IF ELSE and WHILE braces, plus white space cleanup |
ASTERISK-14617: Registration fails when the packe contains 'transport=udp" |
ASTERISK-14618: [patch] Dynamic mapping between peer and extensions using regexten |
ASTERISK-14619: Sip reload fails to erase old peers |
ASTERISK-14620: [patch] dahdi_read unbalanced ast_mutex_lock and ast_mutex_unlock |
ASTERISK-14621: [patch] different 'ringt' timeout counting styles througout chan_dahdi/sig_analog code |
ASTERISK-14622: [patch] WaitForSilence never exits - no dsp.conf |
ASTERISK-14623: [patch] app_queue crashes randomly, it seems to be during call-transfers |
ASTERISK-14624: [patch] Improved penalty use with new ring strategy(penalty ringall), prioritized agents, and status updates. |
ASTERISK-14625: The "port" parameter for an outbound provider is not being respected |
ASTERISK-14626: libc6/malloc/free abort of asterisk |
ASTERISK-14627: [patch] ARG Variables are not overwritten when using empty values on macros |
ASTERISK-14628: [patch] Modify init-scripts for better Fedora compatibility |
ASTERISK-14629: [patch] Use pkgconfig to check for Lua |
ASTERISK-14630: Asterisk Network Connectivity Stops when MySQL Server Not Available |
ASTERISK-14631: [patch] Add empty line after each option in documentation |
ASTERISK-14632: Calls parked via AMI announce to caller instead of callee |
ASTERISK-14633: [patch] Individual maxmessage/maxsecs does not work |
ASTERISK-14634: most cleaner alaw don't compile |
ASTERISK-14635: [patch] If enable DEBUG_FD_LEAKS - h323 can't start. |
ASTERISK-14636: [patch] using ast_free instead of mixmonitor_free |
ASTERISK-14637: After update from 1.4.26 to 1.6.1.4 outgoing call and h323 trace does not work ... |
ASTERISK-14638: [patch] Not possible to specify expiry for peer callback |
ASTERISK-14639: No audio in alerting state when call from Mera to * |
ASTERISK-14640: regression on LAGRQ in chan_iax2 1.6.0.13 |
ASTERISK-14641: [patch] MeetMe privilege escalation in password query |
ASTERISK-14642: Segmentation fault (core dumped) |
ASTERISK-14643: [patch] app_festival hangs on reading from spawned subprocess |
ASTERISK-14644: [patch] Passing the mute flag to MeetMe() makes the new user have "muted" himself, not an admin mute |
ASTERISK-14645: Program terminated with signal 11, Segmentation fault. (retval = (*el->el_map.func[cmdnum]) (el, ch);) |
ASTERISK-14646: segmentation fault when using mixmonitor with two calls |
ASTERISK-14647: Typo in LDAP schema files on line 598 |
ASTERISK-14648: [patch] Only deprecated "rtp debug ip <addr>" works, not "rtp set debug ip <addr>" |
ASTERISK-14649: IAX call received and dialled out to another IAX peer: 100s of CONTROL/ACK/NACK exchanged as call is answered |
ASTERISK-14650: Outgoing DAHDI Calls run to completion without verifying that other end actually answers |
ASTERISK-14651: [patch] Asterisk won't build with curl unless curl_config is present |
ASTERISK-14652: Whisper feature does not work. |
ASTERISK-14653: [patch] chan_sip fails to destroy channels in INVITE when no response received |
ASTERISK-14654: MWI is not sent to a SIP phone upon registration, but is after the mailbox is updated/checked |
ASTERISK-14655: [patch] Distinctive ring detection not working. Exits after first cadence |
ASTERISK-14656: [patch] Crash in ast_readaudio_callback |
ASTERISK-14657: opendir() return code is not checked in last_message_index() |
ASTERISK-14658: Cannot Disable logging of uniqueid |
ASTERISK-14659: 100% CPU with Asterisk 1.4.21.2~dfsg-3 |
ASTERISK-14660: asterisk-addons-1.4.9 does not compile agains libmysqlclient15off (Debian) |
ASTERISK-14661: [patch][regression] set talker detection (T) does not work unless set talker optimization (o) is enabled. |
ASTERISK-14662: [patch] password change for mailboxes without user name |
ASTERISK-14663: [patch] Message Waiting Indication(MWI) is randomly generated when FXO is set to DTMF Caller ID |
ASTERISK-14664: Calls transfered to parking lot immediatly timeout and ring back extension that tried to park call |
ASTERISK-14665: app_voicemail / vm_intro_de fails to play "digits/1" because it tries to say "digits/1F" |
ASTERISK-14666: IMAP greetings not stored in dovecot |
ASTERISK-14667: [patch] CDR dispositions BUSY and FAILED are reported as NO ANSWER |
ASTERISK-14668: ChanSpy's whisper mode is delayed. |
ASTERISK-14669: [patch] manager keeps creating /tmp/ast-ami-XXXXXX files (without deleting) when a single manager client remains logged in |
ASTERISK-14670: Remote crash on Asterisk 1.6.1.4 |
ASTERISK-14671: Analog phone displays unknown caller after hangup |
ASTERISK-14672: [patch] CALLINGSUBADDR incorrectly stated as "Called PRI Subaddress" |
ASTERISK-14673: Asterisk Chrashes Daily |
ASTERISK-14674: [patch] crash: "chan_sip.c", lineno=1934 |
ASTERISK-14675: chan_sip lockup by DNS and connect timeouts |
ASTERISK-14676: [patch] Directory causes crash if dialing by last name |
ASTERISK-14677: dtmf creating double digits if received out of order |
ASTERISK-14678: Set(MONITOR_EXEC= in the queues.conf file doesn't actually do anything |
ASTERISK-14679: Originating g729 SIP channel to DialPlan AGI Requiring Decoding |
ASTERISK-14680: res_config_mysql doesn't recognize commented record correctly |
ASTERISK-14681: 100% CPU after caller hangup when dial are in a loop in a macro |
ASTERISK-14682: r213117 broke registration line parsing |
ASTERISK-14683: Background always returns an AGI result of zero if interrupted by DTMF tone |
ASTERISK-14684: [patch] Core dump in ast_bridge_call features.c line 2772 |
ASTERISK-14685: [patch] Thread debugging version of DEADLOCK_AVOIDANCE: wrong line number if re-lock fails. |
ASTERISK-14686: [patch] Custom devices do not interoperate with distributed events properly |
ASTERISK-14687: Asterisk crashes when voicemail can't access data files |
ASTERISK-14688: Description in queues.conf on call recording is slightly misleading |
ASTERISK-14689: Asterisk 1.6.2 beta 4 has UDP socket leak when using Sip Timers |
ASTERISK-14690: [branch] Add support for distributing device state and MWI via XMPP PubSub |
ASTERISK-14691: Park application expects timeout to be in Milliseconds |
ASTERISK-14692: [patch] Need makefile to do kbuild-like out-of-source-tree builds for Asterisk modules |
ASTERISK-14693: [patch] extension is not recognized in register statement |
ASTERISK-14694: ooh323 fails to register a prefix to a gatekeeper. |
ASTERISK-14695: [patch] Frame.c adjustment for Speex |
ASTERISK-14696: SIP registration with a high latency peer fails |
ASTERISK-14697: [patch] Default Extension (callback) Not Being Set |
ASTERISK-14698: [patch] Incorrect parsing of day range in pbx.c |
ASTERISK-14699: Two consecutive blind transfer crashes Asterisk |
ASTERISK-14700: error on compiling asterisk-addons-1.6.2.0-rc1 |
ASTERISK-14701: SIP qualify goes out of control and kills links |
ASTERISK-14702: [patch] useless message pops hundreds of times per minute |
ASTERISK-14703: [patch] faxing with T.38 fails |
ASTERISK-14704: [patch] New submission guidelines for language additions |
ASTERISK-14705: Automatic progress indication breaks some scenarios |
ASTERISK-14706: MONITOR_FILENAME fails if directory doesnt exist. |
ASTERISK-14707: SendFax without reliable transmission. |
ASTERISK-14708: [patch] AJAM causing Asterisk Seg Fault when attempting login |
ASTERISK-14709: Privacy Manager bugs |
ASTERISK-14710: Large Number of Invites never discarded in sip channels |
ASTERISK-14711: Unexpected Channel selection 3 |
ASTERISK-14712: chan_mobile: audio trouble |
ASTERISK-14713: Queue announcement does not work while using 'r' option |
ASTERISK-14714: Reinvite before channel state is changed to up. |
ASTERISK-14715: [RFC 3389] Voice Activity Detection (VAD) and comfort noise support |
ASTERISK-14716: T38 passthrough errors in Asterisk 1.6.0.14-rc1 |
ASTERISK-14717: OPTIONS sent to 5060 regardless of port specified |
ASTERISK-14718: crash: in "ast_fdleak_fclose" at astfd.c:201 |
ASTERISK-14719: dnsmgr refreshes hostname, but changes SIP port to 0 |
ASTERISK-14720: [patch] Memory leak in res_config_ldap when using realtime |
ASTERISK-14721: TRUNK does not compile on Darwin (MacOS 10.5.8) |
ASTERISK-14722: [Aug 27 13:26:11] NOTICE[28226]: chan_dahdi.c:8703 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 |
ASTERISK-14723: [patch] [regression] Simultaneous calls from same Call-ID silently ignored by asterisk |
ASTERISK-14724: Fails to configure after gtk-config check |
ASTERISK-14725: [patch] asterisk 1.6.2.0-beta4 crash when including nonexistent file from /etc/asterisk/manager.conf |
ASTERISK-14726: failure to to jump to fax extension on fax tone detect |
ASTERISK-14727: [patch] chan_local deadlock |
ASTERISK-14728: Hard crash in send_packet (Asterisk 1.2.33-BRIstuffed-0.3.0-PRE-1y-u) |
ASTERISK-14729: Abort/Raise from try_calling/queue_exec |
ASTERISK-14730: Calleer don´t hear the called channel |
ASTERISK-14731: [patch] sip session timer: Does not work if initial INVITE min-se timer is too small |
ASTERISK-14732: Queue AutoAnswer |
ASTERISK-14733: [patch] Unable to carry over application map functions from 1.4.24 to 1.4.26 |
ASTERISK-14734: [patch] Refcount error in app_queue |
ASTERISK-14735: IAX2 RSA authentication --> calls always unauthenticated |
ASTERISK-14736: asterisk sends no RELEASE_COMPLETE when mISDN Call gets disconnected from asterisk side in 1.6.0.13 and likely others |
ASTERISK-14737: The order of the execution inside the Dial app is wrong |
ASTERISK-14738: [patch] say.conf for french |
ASTERISK-14739: [patch] Fetching SIP headers from BYE sent by callee |
ASTERISK-14740: chan_sip.c : SIP_PAGE2_CALL_ONHOLD* flags missing a bit |
ASTERISK-14741: [patch] chan_sip will not retransmit an ACK |
ASTERISK-14742: Video worked in 1.6.1.4 with SDP patch, but doesn't work in 1.6.1.5 with or without the patch |
ASTERISK-14743: [patch] LDAP error while unregister UA |
ASTERISK-14744: pbx_builtin_background function does not handle fax detection correctly |
ASTERISK-14745: rtt should be stored as double in struct ast_rtp_instance_stats |
ASTERISK-14746: [patch] 100% Cpu usage when caller hangup while app_dial is executing a Macro |
ASTERISK-14747: [patch] callprogress and faxdetect not read when building channel results in fax detection failure |
ASTERISK-14748: Hold Time report (reportholdtime) issue |
ASTERISK-14749: [patch] SoftHangup() incorrectly truncates multi-hyphen channel names |
ASTERISK-14750: [patch] dtmf creating double digits if received out of order |
ASTERISK-14751: In CDR Disposition save NO ANSWER instead BUSY or FAILED |
ASTERISK-14752: ${MEMBERINTERFACE} variable in 1.6.0.x |
ASTERISK-14753: [patch] git-asterisk-howto: document a local git-svn repository of asterisk |
ASTERISK-14754: [patch][regression] LIMIT_TIMEOUT_FILE is not functional |
ASTERISK-14755: Having problems with random crashes |
ASTERISK-14756: crash in local_attended_transfer, likely related to moh - 1.4.26.1 |
ASTERISK-14757: [patch] Backport of function DEVICE_STATE() for Asterisk 1.4 |
ASTERISK-14758: [patch] buggy output in "sip show channelstats" |
ASTERISK-14759: "keepstats" directive mistakenly(?) reverted? |
ASTERISK-14760: Sometimes, callers get stuck in the queue |
ASTERISK-14761: Proper codec not used when picking up a parked call |
ASTERISK-14762: T38 outgoing fax produces RTP error |
ASTERISK-14763: Incoming Only Latency And Jitters every 55 seconds |
ASTERISK-14764: Delayed audio/one leg only |
ASTERISK-14765: SIP register via tls causes lock on sip reload |
ASTERISK-14766: TCP calls no longer bridge (last working version was 1.6.0.1?) |
ASTERISK-14767: [patch] dnsmgr: problem handling A and SRV record changes/problem with multiple A/SRV records returned |
ASTERISK-14768: Update of LDAP not possible when deregistering SIP-Device |
ASTERISK-14769: [patch] queue_log is inconsistent for member information - 2 events use member location where the rest use member name |
ASTERISK-14770: channel variable ${CALLERID(num)} is empty |
ASTERISK-14771: Updateconfig not saving to database |
ASTERISK-14772: [patch] music on hold digit=X not persistent across periodic announcements |
ASTERISK-14773: Transfering phone left connected |
ASTERISK-14774: [patch] iax2 encryption failed on asterisk 1.4.26.2 |
ASTERISK-14775: Compilation - Make install sounds |
ASTERISK-14776: [patch] Tweak to make Music On Hold play files in alphabetical order. |
ASTERISK-14777: reloading doesn't clean everything |
ASTERISK-14778: Please remove IAXy's firmware from the main tarball |
ASTERISK-14779: [patch] caller id number is empty |
ASTERISK-14780: [patch] double free or corruption (!prev) in moh_files_generator |
ASTERISK-14781: Crash when doing chanspy |
ASTERISK-14782: No call progress ringback sent to PSTN caller when Answer() is used in context |
ASTERISK-14783: Talker optimization disables all audio in conference |
ASTERISK-14784: Crash during attended transfer occurs |
ASTERISK-14785: Asterisk is looking in the wrong location for the asterisk.pid (and probably asterisk.ctl) file. |
ASTERISK-14786: [patch] NOTIFY to contains double sip: |
ASTERISK-14787: [patch] segfault when transferring a queue caller |
ASTERISK-14788: Recent change to peer_iphash_cb breaks peer matching |
ASTERISK-14789: [patch] MP3 audio playback is distorted after applying patch for issue 15109 |
ASTERISK-14790: [patch] res_limit.c: refinition of _XOPEN_SOURCE |
ASTERISK-14791: new exten pattern matching not correctly prioritizing extensions by CID |
ASTERISK-14792: [patch] Hints/extension state random in 1.6.0.15 |
ASTERISK-14793: Crash in device state handling |
ASTERISK-14794: [patch] Posibility to send two channels in different direcitons |
ASTERISK-14795: Crash on ael inclusion loop |
ASTERISK-14796: [patch] with 'transport=tls' and host not dynamic, port defaults to 5060 rather than 5061. |
ASTERISK-14797: [patch] musiconhold crash on unload |
ASTERISK-14798: [patch] Revision 152765 introduces regression in stdexten |
ASTERISK-14799: [patch] Revision 152765 introduces scoping difficulties in stdexten |
ASTERISK-14800: [patch][regression] 1.4.26.2 upgrade from 1.4.18 broke NOTIFY keep-alive reponse and stale nonce handling to Linksys SPA962 |
ASTERISK-14801: wrong parsing of received RTCP packets |
ASTERISK-14802: Responce for Action 'Ping' do not complete in tcp stream |
ASTERISK-14803: chan_sip deadlock in mutex sip_alloc |
ASTERISK-14804: BroadVoice With Asterisk |
ASTERISK-14805: idle odbc connections are not cleaned up |
ASTERISK-14806: Crash on exit |
ASTERISK-14807: Corrupt Memory Issue - with Valgrind Trace |
ASTERISK-14808: [patch] SIPshowregistry manager action obmits ActionID from RegistryEntry events |
ASTERISK-14809: [patch] 1.4.26.2 still builds with freeplay MOH sounds |
ASTERISK-14810: [patch] res_config_odbc.c compilation fix for non-Unicode UnixODBC versions |
ASTERISK-14811: [patch] Channels that was placed in "meetme" in dialing state do not leave application in CONGESTION case |
ASTERISK-14812: Asterisk crash asterisk atxfer to a queue |
ASTERISK-14813: [Patch] Asterisk updated LDAP Schema |
ASTERISK-14814: Interoperability with Exchange 2007 UM |
ASTERISK-14815: [patch] First caller to dynamic conference has to enter pin number twice |
ASTERISK-14816: [patch] voicemail should mark messages as read by doing an UPDATE instead of an INSERT followed by a DELETE |
ASTERISK-14817: [patch] Contact header port ignores transport when using externip |
ASTERISK-14818: asterisk does not send by "BYE" to sip peer |
ASTERISK-14819: Dial() with H drops inbound call |
ASTERISK-14820: NewChannel AMI event on DAHDI (or Zaptel) channels contains CallerID information from previous call |
ASTERISK-14821: Recording Stops After Transfer from Queue |
ASTERISK-14822: T38 udptl.c bufferoverflow |
ASTERISK-14823: SEGV in chan_iax2.c - socket_process - with Zoiper client |
ASTERISK-14824: problem with call file data entry in Asterisk CDR Database |
ASTERISK-14825: crash because of invalid cdr->dst string |
ASTERISK-14826: 1.6.1.5 - "Ghost" channels |
ASTERISK-14827: problem with call file data entry in Asterisk CDR Database |
ASTERISK-14828: Memory leak on reload_config |
ASTERISK-14829: [patch] When useragent was added to realtime_update_peer in chan_sip the necessary ldap changes were not made |
ASTERISK-14830: [patch] handle_response check for retransmits when using TCP/TLS |
ASTERISK-14831: SIP/TCP always shows registered peer as UNREACHABLE |
ASTERISK-14832: Crash in ast_variable_new (config.c) |
ASTERISK-14833: crash on second 'dahdi destroy channel' if a var was set |
ASTERISK-14834: Console flood & CPU load 100% when IAX2 channel falls |
ASTERISK-14835: Broken audio with g.722 if jitterbuffer is used. |
ASTERISK-14836: [patch] features.conf auto-included park function sometimes included junk in Park() command |
ASTERISK-14837: [patch] Document to describe workflow of Asterisk open source issue tracker |
ASTERISK-14838: udptl.conf parameter T38FaxUdpEC is not working correctly if set to t38UDPRedundancy |
ASTERISK-14839: [patch] Regression: 1.6.0 fixes for no incoming calls, memory leak, and hosts file warning were not merged into 1.6.1 |
ASTERISK-14840: [patch] TCP/TLS invites(and possibly others) broken from r218504 and onward |
ASTERISK-14841: rtptimeout option doesn't work for inbound calls |
ASTERISK-14842: [patch] Add ability to log CONGESTION calls to CDR |
ASTERISK-14843: When no callerid is recieved, cannot override callerid |
ASTERISK-14844: [patch] Deadlock in channel masquerade handling |
ASTERISK-14845: [patch] chan_h323 with h323plus 1.21.0 doesn't compile |
ASTERISK-14846: auto-loading res_snmp causes Asterisk to Seg fault |
ASTERISK-14847: Monitor does not produce output files (via dial plan or queues) |
ASTERISK-14848: crash when calling ao2_unlock inside pthread_timer_disable_continuous |
ASTERISK-14849: SIP Realtime appears to be completely broken |
ASTERISK-14850: Voicemails fail to be stored in IMAP |
ASTERISK-14851: Music On Hold |
ASTERISK-14852: Asterisk generates a BYE after 15 minutes or more consistently on trunk calls |
ASTERISK-14853: phoneprov route variable values missing |
ASTERISK-14854: [patch] Crash when freeing buffer in update_curl |
ASTERISK-14855: pulse and hardware dtmf digits not detected in chan_dahdi/trunk |
ASTERISK-14856: New sounds for sounds-extras |
ASTERISK-14857: 'meetme list' doesn't list conferences, shows usage only |
ASTERISK-14858: billsec not 0 for NO ANSWER calls |
ASTERISK-14859: [patch] CLI to honor user's ~/.editrc file |
ASTERISK-14860: Modifieied CALLERID(num) is not logged into CDRs as expected |
ASTERISK-14861: [patch] Make internal_timing on by default |
ASTERISK-14862: Issue in Blind Transfer |
ASTERISK-14863: [patch] app_voicemail.so doesn't refresh information from database on reload |
ASTERISK-14864: Faulty SIP device caused not just asterisk but the entire system to lock up every 20 minutes on the DOT! |
ASTERISK-14865: Problem with call capture |
ASTERISK-14866: Variables set inside a macro lose their value after dial |
ASTERISK-14867: memory leak, tcptls_session never destroyed in chan_sip for client connections |
ASTERISK-14868: [patch] IAX does not allow CALLERID(num) contain non-numbers. |
ASTERISK-14869: billsec not available in "h" extensions |
ASTERISK-14870: Asterisk does not use the “expires=” from the SIP contact header during registration |
ASTERISK-14871: Registration against a SIP provider fails in 1.6.2.0 |
ASTERISK-14872: [patch] if caller ID name is not set dahdi crashes |
ASTERISK-14873: Phone number change in doc/lang/language-criteria.txt |
ASTERISK-14874: RemoveQueueMember can't remove from all queues |
ASTERISK-14875: Asterisk crashes when playing back voicemail that has been forwarded with a prepend |
ASTERISK-14876: [patch] correct auth keyword parsing in add_realm_authentication() |
ASTERISK-14877: Options flood to sip user, after adding dynamic a hint |
ASTERISK-14878: core show locks stop responding and crash asterisk |
ASTERISK-14879: Sometimes asterisk Crashes in the hangup phase of misdn |
ASTERISK-14880: [patch] sqlite3 CDRs not working after a reload |
ASTERISK-14881: Mixmonitor stop recording after atxfer |
ASTERISK-14882: Asterisk dies with error code 127 |
ASTERISK-14883: Create contrib/realtime/{mysql,pgsql,oracle,mssql}/<family>.sql files for realtime |
ASTERISK-14884: [patch] Not recording the duration of transfers. |
ASTERISK-14885: Core show hints report wrong state |
ASTERISK-14886: Asterisk crashed after transfer |
ASTERISK-14887: [patch] SQLColumns Error When Using Postgresql Schema |
ASTERISK-14888: Conference room with app ConfBridge has no audio. |
ASTERISK-14889: [patch] Deadlock in ChanSpy |
ASTERISK-14890: Asterisk generates BYE at EXACTLY 900 seconds (15 min) and terminates call |
ASTERISK-14891: [patch] Asterisk ignores changes to realtime queue member table after initial startup |
ASTERISK-14892: [patch] Best practices for programming in C |
ASTERISK-14893: chan_h323 and asterisk-1.6.2 compile issue |
ASTERISK-14894: [patch] Support for physical tapping in PRI lines |
ASTERISK-14895: [patch] Dropping frame since I'm still dialing on Zap/... (resp. DAHDi/...) with DIGITAL calls |
ASTERISK-14896: [patch] Huge memory consumption after few hours of load |
ASTERISK-14897: Explicitly set CallerID is lost |
ASTERISK-14898: Unable to change the packetization settings (ptime) for codecs from default of 20ms |
ASTERISK-14899: abort in filestream_destructor / ast_filestream_frame_freed / moh_files_generator |
ASTERISK-14900: automon "*1" is recognized but does not write any file |
ASTERISK-14901: Voicemail MWI - deleted messages are not notified to the extension |
ASTERISK-14902: QueueMemberState events only sent when phone rebooted |
ASTERISK-14903: [patch] implement manager command DAHDIShowSpans |
ASTERISK-14904: [patch] Segmentation fault in queue_cmp_cb |
ASTERISK-14905: SIP response 302 "Moved Temporarily" change the Language setting to "en" |
ASTERISK-14906: QUEUE_MEMBER and QUEUE_MEMBER_COUNT tries to destroy queue, leading to segmentation fault |
ASTERISK-14907: chan_h323 and ptlib-2.4.5 and h323plus-1.21.1 |
ASTERISK-14908: ao2_iterator_init() does not hold a reference to the container it is iterating |
ASTERISK-14909: Finish implementation of OBJ_MULTIPLE support in astobj2 |
ASTERISK-14910: [patch] Added mohsuggest info to output for CLI: sip show peer {name} |
ASTERISK-14911: [patch] Channel reference leak when calling through an optimized local channel |
ASTERISK-14912: [patch] Asterisk version 1.6.1.6 |
ASTERISK-14913: Segfault in sip_send_mwi_to_peer |
ASTERISK-14914: Wrong handling of INVITE with Diversion tag when Asterisk has seen the callid before |
ASTERISK-14915: [patch] change the order for ACF registration for SMDI |
ASTERISK-14916: [patch] segfault in 1.6.1.6 in _ao2_find, called from chan_iax2 after approx. 75.000 calls |
ASTERISK-14917: [patch] Reset entire span request can result in a crash |
ASTERISK-14918: [patch] Strange nasty sound (Because Asterisk tryes to handle new voicemail, but there is no voicemails, voicemail isn't used) |
ASTERISK-14919: [patch] Schema problem with res_pgsql |
ASTERISK-14920: schema problem |
ASTERISK-14921: FreeTDS not detected by Asterisk 1.4.26 |
ASTERISK-14922: "core show channels" crash |
ASTERISK-14923: Bug 0014309 for 1.4 has been reproduced on my 1.6.1 system |
ASTERISK-14924: [patch] Call does not drop when caller hangs up |
ASTERISK-14925: [patch] call time limit overflow |
ASTERISK-14926: [patch] Clean valgrind output by suppressing false errors |
ASTERISK-14927: chan_alsa.c: snd_pcm_open failed: No such file or directory |
ASTERISK-14928: [patch] Exten channel request hangup to accept "all" and hangup all channels |
ASTERISK-14929: Call abort after wrong behaviour of VNAK transmission |
ASTERISK-14930: [patch] Crash on unregister SIP realtime peer - double free |
ASTERISK-14931: Asterisk forks processes then locks up |
ASTERISK-14932: [patch] SIP CHANNEL(rtpqos,audio,...) variables missing. |
ASTERISK-14933: [patch] crash in ast_frame_free / ast_generic_bridge |
ASTERISK-14934: Asterisk cuts audio to the internal extension |
ASTERISK-14935: [regression] menuselect compilation failure on Solaris 10 |
ASTERISK-14936: fullcontact not updated in database |
ASTERISK-14937: v23 caller id detection (UK) less than 100% |
ASTERISK-14938: Asterisk 1.6.2 RC2: func_curl crashes Asterisk |
ASTERISK-14939: AEL parsers does not find existing label |
ASTERISK-14940: [patch] Warnings during configuration |
ASTERISK-14941: [patch] sippeers loaded with realtime are treated as type=friends, no matter what type is in the db |
ASTERISK-14942: [patch] App_jack.so JACK_HOOK half works |
ASTERISK-14943: support hints in pbx_lua |
ASTERISK-14944: [patch] failed to negiotate t38 |
ASTERISK-14945: CDR blind transfer problem |
ASTERISK-14946: [patch] Crash on Pickup or Transfer |
ASTERISK-14947: [patch] Polycom 000000000000.cfg template uses old parameter separator |
ASTERISK-14948: conf2ael segfaults if the following code is in my extensions.conf |
ASTERISK-14949: [patch] sip peers loaded with realtime doesnt load useragent |
ASTERISK-14950: Missing routing info for voicemail notifications |
ASTERISK-14951: AMI redirect of caller channel while being answered by queue agent causes caller disconnect |
ASTERISK-14952: [patch] AGI returns bogus "510 Invalid or unknown command" |
ASTERISK-14953: [patch] Autocreated peers not deleted when unregister explicitly, become zombies |
ASTERISK-14954: [patch] Status of dahdi/zap channels incorrectly reported unavailable instead of idle |
ASTERISK-14955: Asterisk unable to bridge RTP when a peer server performs a call transfer when canreinvite is enabled |
ASTERISK-14956: Realtime Queue Crashes Asterisk |
ASTERISK-14957: Various valgrind errors in 1.6.0.16-rc2 |
ASTERISK-14958: SendFax 'a' option |
ASTERISK-14959: [patch] reference argument sub in handle_transfer_button after we check it |
ASTERISK-14960: [patch] Asterisk 1.4.27-rc2 crash |
ASTERISK-14961: [patch] Paused members in queue with higher weight |
ASTERISK-14962: blf not working on aastra 57i/grandstream |
ASTERISK-14963: [patch] crash when spying - was working fine in beta2 |
ASTERISK-14964: [patch] crash in ast_rtp_instance_early_bridge_make_compatible() when directmedia=yes |
ASTERISK-14965: app_voicemail does not load in 1.4.25.1 and 1.4.26.2 and 1.4.27rc2 |
ASTERISK-14966: [patch] ast_transfer will stall until hangup if called with a channel that doesn't support transfers. |
ASTERISK-14967: [patch] Call fails to go through starting with build 1.6.0.14 |
ASTERISK-14968: Fax detection broken when upgrading from 1.6.0.9 to 1.6.0.15 |
ASTERISK-14969: how to support DTMF type caller id on FXO port? |
ASTERISK-14970: Shell Function is missing and very needed in all editions. |
ASTERISK-14971: [patch] Crash on deeply nested while/if statements in AEL |
ASTERISK-14972: Asterisk 1.6.1.6 not closing RTP ports after connection |
ASTERISK-14973: Loss of IAX calls after strange VNAK behavior |
ASTERISK-14974: Lenth of called number send in enblock |
ASTERISK-14975: [patch] Asterisk crashes with "Fixup failed on channel XXX, strange things may happen." |
ASTERISK-14976: [patch] Crash in local_ast_moh_start / ast_indicate_data due to AST_CONTROL_HOLD with bad pointer |
ASTERISK-14977: [patch] Incorrect parsing of 'hint' extensions |
ASTERISK-14978: [patch] Peer mismatch in incomming call |
ASTERISK-14979: [patch] Asterisk crashes after "core stop gracefully" |
ASTERISK-14980: [patch] README has old help information |
ASTERISK-14981: channel.c:2711 __ast_read: Exception flag set on 'SIP/xxx', but no exception handler |
ASTERISK-14982: Field "called_party" is missing when making outbound H.323 calls |
ASTERISK-14983: [bounty] Redirect Caller After Hangup from Queue |
ASTERISK-14984: Crash on incoming IAX call from Asterisk 1.4.18 without DONT_OPTIMIZE |
ASTERISK-14985: [patch] Seg Fault on Transfer with IAX/iLBC |
ASTERISK-14986: [patch] Warning log message with debug info |
ASTERISK-14987: AMI input stream limit |
ASTERISK-14988: [patch] Not all SIP extensions receive a page |
ASTERISK-14989: problem with waiting queue |
ASTERISK-14990: [patch] Asterisk will never retry after the first register to H.323 gk fails. |
ASTERISK-14991: AGI not destroyed when got hangup while Get Data or Get Option command |
ASTERISK-14992: [patch] length of queue name is static |
ASTERISK-14993: bridged zaptel channel issue |
ASTERISK-14994: [patch] utils.c:938 ast_carefulwrite: Timed out trying to write causes corruption to astdb |
ASTERISK-14995: Dahdi not compile on kernel-2.6.31 |
ASTERISK-14996: CLI incorrectly delivered when there is no calling number |
ASTERISK-14997: Asterisk MOH playing old audio for first 30 to 60 seconds |
ASTERISK-14998: Crash in __ast_pthread_mutex_unlock chan_sip after park |
ASTERISK-14999: No streaming musiconhold when using dial command |